A digital microphone converts an acoustic pressure wave to a digital audio signal. The output of the digital microphone is usually processed by a separate codec device or another separate audio signal processing system. Thus, depending on specifications of different codec devices and audio signal processing systems, the digital microphone may to work at a variety of clock rates/sampling frequencies, and the output signal stream of the digital microphone may support different rates and formats. The digital microphone may also be compact (or area efficient) and power efficient in order for it to be integrated with other devices, especially when being used by today's power sensitive embedded devices and mobile applications. These features bring challenge to a conventional digital microphone where the whole system has to be designed for the highest clock rate.
The conventional design may not be power efficient because key electronic components of a digital microphone (e.g., a digital filter, an analog-to-digital converter (ADC), and a digital modulator, etc.) may consume more power when working at higher frequencies. For instance, as the frequency of the oversampled ADC increases, there is a corresponding decrease in the settling time of the ADC. In some cases, more power is consumed to effect the settling time. In addition, characteristics of a digital filter may vary depending on the sampling frequency the filter operates at, which either affects the performance of the digital filter when switching to a different sampling frequency, or uses an adaptation of programming coefficients of the digital filters. Therefore, an efficient digital microphone implementation topology for supporting multiple sampling frequencies is desired.